diff --git a/README b/README
new file mode 100644
index 0000000..ca3b76d
--- /dev/null
+++ b/README
@@ -0,0 +1,50 @@
+PyAlsaAudio
+===========
+
+Author: Casper Wilstrup (cwi@unispeed.dk)
+
+This package contains wrappers for acessing the ALSA api from Python. It
+is currently fairly complete for PCM devices. My next goal is to have
+complete mixer supports as well. MIDI sequencer support is low on my
+priority list, but volunteers are welcome.
+
+If you find bugs in the wrappers please notify me on email. Please
+don't send bug reports regarding ALSA specifically. There are several
+bugs in this api, and those should be reported to the ALSA team - not
+me.
+
+This software is licensed under the PSF license - the same one used
+by the majority of the python distribution. Basically you can use it
+for anything you wish (even commercial purposes). There is no warranty
+whatsoever.
+
+
+Installation
+============
+
+Note: the wrappers link with the alsasound library alsa (from the alsa-lib
+package). Verify that this is installed by looking for /usr/lib/libasound.so
+before building. Naturally you also need to use a kernel with proper ALSA
+support. This is the default in Linux kernel 2.6 and later. If you are using
+kernel version 2.4 you may need to install the ALSA patches yourself - althoug
+most distributions ship with ALSA kernels.
+
+To install, execute the following:
+  $ python setup.py build
+
+And then as root:
+  # python setup.py install
+
+
+Using the API
+=============
+
+There are two example programs included with the source:
+'playbacktest.py' which plays back raw sound data read from
+stdin
+
+'recordtest.py' which captures sound from the microphone at writes
+it raw to stdout.
+
+In a future version, I'll include documentation with the package, but
+for now, you're stuck with the examples.
diff --git a/TODO b/TODO
new file mode 100644
index 0000000..e12a0b5
--- /dev/null
+++ b/TODO
@@ -0,0 +1,3 @@
+- Implement the Mixer API
+
+- Implement MIDI/sequencer support.
diff --git a/alsaaudio.c b/alsaaudio.c
new file mode 100644
index 0000000..6086c43
--- /dev/null
+++ b/alsaaudio.c
@@ -0,0 +1,459 @@
+/*
+ * alsaaudio -- Python interface to ALSA (Advanced Linux Sound Architecture).
+ *              The standard audio API for Linux since kernel 2.0
+ *
+ * Contributed by Unispeed A/S (http://www.unispeed.com)
+ * Primary author: Casper Wilstup (cwi@unispeed.dk)
+ *
+ *
+ * (c) 2004 by Unispeed A/S. All rights reserved.
+ *
+ * License: CRNI's Python 1.6 license (or any later version of the same)
+ *
+ */
+
+#include "Python.h"
+#include <alsa/asoundlib.h>
+#include <stdio.h>
+
+
+typedef struct {
+  PyObject_HEAD;
+  int pcmtype;
+  int pcmmode;
+  char *pcmname;
+  
+  snd_pcm_t *handle;
+
+  // Configurable parameters
+  int channels;
+  int rate;
+  int format;
+  snd_pcm_uframes_t periodsize;
+  int framesize;
+
+} alsapcm_t;
+
+static PyTypeObject ALSAAudioType;
+
+static PyObject *ALSAAudioError;
+
+static int alsapcm_setup(alsapcm_t *self) {
+  int res,dir;
+  unsigned int val;
+  snd_pcm_hw_params_t *hwparams;
+  snd_pcm_uframes_t frames;
+
+  if (self->handle) {
+    snd_pcm_close(self->handle);
+    self->handle = 0;
+  }
+  res = snd_pcm_open(&(self->handle),self->pcmname,self->pcmtype,self->pcmmode);
+  if (res < 0) return res;
+
+  snd_pcm_hw_params_alloca(&hwparams);
+  res = snd_pcm_hw_params_any(self->handle, hwparams);
+  if (res < 0) return res;
+
+  snd_pcm_hw_params_alloca(&hwparams);
+
+  /* Fill it in with default values. */
+  snd_pcm_hw_params_any(self->handle, hwparams);
+  snd_pcm_hw_params_set_access(self->handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
+  snd_pcm_hw_params_set_format(self->handle, hwparams, self->format);
+  snd_pcm_hw_params_set_channels(self->handle, hwparams, self->channels);
+  dir = 0;
+  snd_pcm_hw_params_set_rate(self->handle, hwparams, self->rate, dir);
+  snd_pcm_hw_params_set_period_size(self->handle, hwparams, self->periodsize, dir);
+  snd_pcm_hw_params_set_periods(self->handle,hwparams,4,0);
+
+  res = snd_pcm_hw_params(self->handle, hwparams);
+
+  self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;
+  return res;
+}
+
+/* ALSA PCM Object members */
+static alsapcm_t *
+newalsapcmobject(PyObject *args) {
+  int res;
+  alsapcm_t *self;
+  int pcmtype=0;
+  int pcmmode=0;
+  char *pcmname = "default";
+  if (!PyArg_ParseTuple(args,"|iis",&pcmtype,&pcmmode,&pcmname)) return NULL;
+  if (!(self = (alsapcm_t *)PyObject_New(struct alsa_audio, &ALSAAudioType))) return NULL;
+
+  if (pcmtype != SND_PCM_STREAM_PLAYBACK && pcmtype != SND_PCM_STREAM_CAPTURE) {
+    PyErr_SetString(ALSAAudioError, "PCM type must be PCM_PLAYBACK (0) or PCM_CAPTUPE (1)");
+    return NULL;
+  }
+  if (pcmmode < 0 || pcmmode > SND_PCM_ASYNC) {
+    PyErr_SetString(ALSAAudioError, "Invalid PCM mode");
+    return NULL;
+  }
+  self->pcmtype = pcmtype;
+  self->pcmmode = pcmmode;
+  self->pcmname = strdup(pcmname);
+  
+  self->channels = 2;
+  self->rate = 44100;
+  self->format = SND_PCM_FORMAT_S16_LE;
+  self->periodsize = 32;
+
+  self->handle = 0;
+  res = alsapcm_setup(self);
+  if (res < 0) {
+    if (self->handle) {
+      snd_pcm_close(self->handle);
+      self->handle = 0;
+    }
+    PyErr_SetString(ALSAAudioError, snd_strerror(res));
+    return NULL;    
+  }
+  return self;
+}
+
+static void alsapcm_dealloc(alsapcm_t *self) {
+  if (self->handle) {
+    snd_pcm_drain(self->handle);
+    snd_pcm_close(self->handle);
+  }
+  PyObject_Del(self);
+}
+
+static PyObject *
+alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
+  unsigned int val,val2;
+  int dir;
+  snd_pcm_uframes_t frames;
+  snd_pcm_hw_params_t *hwparams;
+  snd_pcm_hw_params_alloca(&hwparams);
+  snd_pcm_hw_params_current(self->handle,hwparams);
+
+
+  if (!PyArg_ParseTuple(args,"")) return NULL;
+
+  printf("PCM handle name = '%s'\n", snd_pcm_name(self->handle));
+  printf("PCM state = %s\n", snd_pcm_state_name(snd_pcm_state(self->handle)));
+
+  snd_pcm_hw_params_get_access(hwparams, (snd_pcm_access_t *) &val);
+  printf("access type = %s\n", snd_pcm_access_name((snd_pcm_access_t)val));
+
+  snd_pcm_hw_params_get_format(hwparams, &val);
+  printf("format = '%s' (%s)\n", 
+	 snd_pcm_format_name((snd_pcm_format_t)val),
+	 snd_pcm_format_description((snd_pcm_format_t)val));
+
+  snd_pcm_hw_params_get_subformat(hwparams, (snd_pcm_subformat_t *)&val);
+  printf("subformat = '%s' (%s)\n",
+	 snd_pcm_subformat_name((snd_pcm_subformat_t)val),
+	 snd_pcm_subformat_description((snd_pcm_subformat_t)val));
+
+  snd_pcm_hw_params_get_channels(hwparams, &val);
+  printf("channels = %d\n", val);
+
+  snd_pcm_hw_params_get_rate(hwparams, &val, &dir);
+  printf("rate = %d bps\n", val);
+
+  snd_pcm_hw_params_get_period_time(hwparams, &val, &dir);
+  printf("period time = %d us\n", val);
+
+  snd_pcm_hw_params_get_period_size(hwparams, &frames, &dir);
+  printf("period size = %d frames\n", (int)frames);
+
+  snd_pcm_hw_params_get_buffer_time(hwparams, &val, &dir);
+  printf("buffer time = %d us\n", val);
+
+  snd_pcm_hw_params_get_buffer_size(hwparams, (snd_pcm_uframes_t *) &val);
+  printf("buffer size = %d frames\n", val);
+
+  snd_pcm_hw_params_get_periods(hwparams, &val, &dir);
+  printf("periods per buffer = %d frames\n", val);
+
+  snd_pcm_hw_params_get_rate_numden(hwparams, &val, &val2);
+  printf("exact rate = %d/%d bps\n", val, val2);
+
+  val = snd_pcm_hw_params_get_sbits(hwparams);
+  printf("significant bits = %d\n", val);
+
+  snd_pcm_hw_params_get_tick_time(hwparams, &val, &dir);
+  printf("tick time = %d us\n", val);
+
+  val = snd_pcm_hw_params_is_batch(hwparams);
+  printf("is batch = %d\n", val);
+
+  val = snd_pcm_hw_params_is_block_transfer(hwparams);
+  printf("is block transfer = %d\n", val);
+
+  val = snd_pcm_hw_params_is_double(hwparams);
+  printf("is double = %d\n", val);
+
+  val = snd_pcm_hw_params_is_half_duplex(hwparams);
+  printf("is half duplex = %d\n", val);
+
+  val = snd_pcm_hw_params_is_joint_duplex(hwparams);
+  printf("is joint duplex = %d\n", val);
+
+  val = snd_pcm_hw_params_can_overrange(hwparams);
+  printf("can overrange = %d\n", val);
+
+  val = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams);
+  printf("can mmap = %d\n", val);
+
+  val = snd_pcm_hw_params_can_pause(hwparams);
+  printf("can pause = %d\n", val);
+
+  val = snd_pcm_hw_params_can_resume(hwparams);
+  printf("can resume = %d\n", val);
+
+  val = snd_pcm_hw_params_can_sync_start(hwparams);
+  printf("can sync start = %d\n", val);
+
+  Py_INCREF(Py_None);
+  return Py_None;
+    
+}
+
+static PyObject *
+alsapcm_pcmtype(alsapcm_t *self, PyObject *args) {
+  if (!PyArg_ParseTuple(args,"")) return NULL;
+  return PyInt_FromLong(self->pcmtype);
+}
+
+static PyObject *
+alsapcm_pcmmode(alsapcm_t *self, PyObject *args) {
+  if (!PyArg_ParseTuple(args,"")) return NULL;
+  return PyInt_FromLong(self->pcmmode);
+}
+
+static PyObject *
+alsapcm_pcmname(alsapcm_t *self, PyObject *args) {
+  if (!PyArg_ParseTuple(args,"")) return NULL;
+  return PyString_FromString(self->pcmname);
+}
+
+static PyObject *
+alsapcm_setchannels(alsapcm_t *self, PyObject *args) {
+  int channels;
+  int res;
+  if (!PyArg_ParseTuple(args,"i",&channels)) return NULL;
+  self->channels = channels;
+  res = alsapcm_setup(self);
+  if (res < 0) {
+    PyErr_SetString(ALSAAudioError, snd_strerror(res));
+    return NULL;    
+  }
+  return PyInt_FromLong(channels);
+}
+
+static PyObject *
+alsapcm_setrate(alsapcm_t *self, PyObject *args) {
+  int rate;
+  int res;
+  if (!PyArg_ParseTuple(args,"i",&rate)) return NULL;
+  self->rate = rate;
+  res = alsapcm_setup(self);
+  if (res < 0) {
+    PyErr_SetString(ALSAAudioError, snd_strerror(res));
+    return NULL;    
+  }
+  return PyInt_FromLong(self->rate);
+}
+
+static PyObject *
+alsapcm_setformat(alsapcm_t *self, PyObject *args) {
+  int format;
+  int res;
+  if (!PyArg_ParseTuple(args,"i",&format)) return NULL;
+  self->format = format;
+  res = alsapcm_setup(self);
+  if (res < 0) {
+    PyErr_SetString(ALSAAudioError, snd_strerror(res));
+    return NULL;    
+  }
+  return PyInt_FromLong(self->format);
+}
+
+static PyObject *
+alsapcm_setperiodsize(alsapcm_t *self, PyObject *args) {
+  int periodsize;
+  int res;
+  if (!PyArg_ParseTuple(args,"i",&periodsize)) return NULL;
+  self->periodsize = periodsize;
+  res = alsapcm_setup(self);
+  if (res < 0) {
+    PyErr_SetString(ALSAAudioError, snd_strerror(res));
+    return NULL;    
+  }
+  return PyInt_FromLong(self->periodsize);
+}
+
+
+static PyObject *
+alsapcm_read(alsapcm_t *self, PyObject *args) {
+  int res;
+  int val,dir;
+  char buffer[8000];
+  snd_pcm_hw_params_t *hwparams;
+
+  if (self->framesize * self->periodsize > 8000) {
+    PyErr_SetString(ALSAAudioError,"Capture data too large. Try decreasing period size");
+    return NULL;
+  }
+
+  if (!PyArg_ParseTuple(args,"")) return NULL;
+  if (self->pcmtype != SND_PCM_STREAM_CAPTURE) {
+    PyErr_SetString(ALSAAudioError,"Cannot read from playback PCM");
+    return NULL;
+  }
+
+  res = snd_pcm_readi(self->handle, buffer, self->periodsize);
+  if (res == -EPIPE) {
+    /* EPIPE means overrun */
+    snd_pcm_prepare(self->handle);
+  }
+  else if (res == -EAGAIN) {
+    res = 0;
+  }
+  else if (res < 0) {
+    PyErr_SetString(ALSAAudioError,snd_strerror(res));
+    return NULL;
+  } 
+
+  return Py_BuildValue("is#",res,buffer,res*self->framesize);
+}
+
+static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
+  char *data;
+  int datalen;
+  int res;
+  if (!PyArg_ParseTuple(args,"s#",&data,&datalen)) return NULL;
+  if (datalen%self->framesize) {
+    PyErr_SetString(ALSAAudioError,"Data size must be a multipla of framesize");
+    return NULL;
+  }
+  res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
+  if (res == -EPIPE) {
+    /* EPIPE means underrun */
+    snd_pcm_prepare(self->handle);
+    snd_pcm_writei(self->handle, data, datalen/self->framesize);
+    snd_pcm_writei(self->handle, data, datalen/self->framesize);
+  }
+  else if (res == -EAGAIN) {
+    return PyInt_FromLong(0);
+  }
+  else if (res < 0) {
+    PyErr_SetString(ALSAAudioError,snd_strerror(res));
+    return NULL;
+  }  
+  return PyInt_FromLong(datalen);
+}
+
+/* Module functions */
+
+static PyObject *
+alsa_openpcm(PyObject *self, PyObject *args) {
+  return (PyObject *)newalsapcmobject(args);
+}
+
+
+/* ALSA PCM Object Bureaucracy */
+
+static PyMethodDef alsapcm_methods[] = {
+  {"pcmtype", (PyCFunction)alsapcm_pcmtype, METH_VARARGS},
+  {"pcmmode", (PyCFunction)alsapcm_pcmmode, METH_VARARGS},
+  {"pcmname", (PyCFunction)alsapcm_pcmname, METH_VARARGS},
+  {"setchannels", (PyCFunction)alsapcm_setchannels, METH_VARARGS},
+  {"setrate", (PyCFunction)alsapcm_setrate, METH_VARARGS},
+  {"setformat", (PyCFunction)alsapcm_setformat, METH_VARARGS},
+  {"setperiodsize", (PyCFunction)alsapcm_setperiodsize, METH_VARARGS},
+
+  {"dumpinfo", (PyCFunction)alsapcm_dumpinfo, METH_VARARGS},
+
+  {"read", (PyCFunction)alsapcm_read, METH_VARARGS},
+  {"write", (PyCFunction)alsapcm_write, METH_VARARGS},
+
+  {NULL, NULL}
+};
+
+static PyObject *
+alsapcm_getattr(alsapcm_t *self, char *name) {
+  return Py_FindMethod(alsapcm_methods, (PyObject *)self, name);
+}
+
+static PyTypeObject ALSAAudioType = {
+  PyObject_HEAD_INIT(&PyType_Type)
+  0,
+  "alsaaudio.pcm",
+  sizeof(alsapcm_t),
+  0,
+  /* methods */
+  (destructor) alsapcm_dealloc, 
+  0,
+  (getattrfunc)alsapcm_getattr,
+  0,
+  0,
+  0,
+};
+
+
+
+/* Initialization */
+
+static PyMethodDef alsaaudio_methods[] = {
+  { "openpcm", alsa_openpcm, METH_VARARGS },
+  { 0, 0 },
+};
+
+#define _EXPORT_CONSTANT(mod, name) \
+  if (PyModule_AddIntConstant(mod, #name, (long) (name)) == -1) return;
+#define _EXPORT_INT(mod, name, value) \
+  if (PyModule_AddIntConstant(mod, name, (long) value) == -1) return;
+
+void initalsaaudio(void) {
+  PyObject *m;
+  m = Py_InitModule("alsaaudio",alsaaudio_methods);
+
+  ALSAAudioError = PyErr_NewException("alsaaudio.ALSAAudioError", NULL, NULL);
+  if (ALSAAudioError) {
+    /* Each call to PyModule_AddObject decrefs it; compensate: */
+    Py_INCREF(ALSAAudioError);
+    Py_INCREF(ALSAAudioError);
+    PyModule_AddObject(m, "error", ALSAAudioError);
+    PyModule_AddObject(m, "ALSAAudioError", ALSAAudioError);
+  }
+
+
+  _EXPORT_INT(m,"PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK);
+  _EXPORT_INT(m,"PCM_CAPTURE",SND_PCM_STREAM_CAPTURE);
+
+  _EXPORT_INT(m,"PCM_NONBLOCK",SND_PCM_NONBLOCK);
+  _EXPORT_INT(m,"PCM_ASYNC",SND_PCM_ASYNC);
+
+  /* PCM Formats */
+  _EXPORT_INT(m,"PCM_FORMAT_S8",SND_PCM_FORMAT_S8);
+  _EXPORT_INT(m,"PCM_FORMAT_U8",SND_PCM_FORMAT_U8);
+  _EXPORT_INT(m,"PCM_FORMAT_S16_LE",SND_PCM_FORMAT_S16_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_S16_BE",SND_PCM_FORMAT_S16_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_U16_LE",SND_PCM_FORMAT_U16_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_U16_BE",SND_PCM_FORMAT_U16_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_S24_LE",SND_PCM_FORMAT_S24_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_S24_BE",SND_PCM_FORMAT_S24_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_U24_LE",SND_PCM_FORMAT_U24_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_U24_BE",SND_PCM_FORMAT_U24_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_S32_LE",SND_PCM_FORMAT_S32_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_S32_BE",SND_PCM_FORMAT_S32_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_U32_LE",SND_PCM_FORMAT_U32_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_U32_BE",SND_PCM_FORMAT_U32_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_FLOAT_LE",SND_PCM_FORMAT_FLOAT_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_FLOAT_BE",SND_PCM_FORMAT_FLOAT_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_FLOAT64_LE",SND_PCM_FORMAT_FLOAT64_LE);
+  _EXPORT_INT(m,"PCM_FORMAT_FLOAT64_BE",SND_PCM_FORMAT_FLOAT64_BE);
+  _EXPORT_INT(m,"PCM_FORMAT_MU_LAW",SND_PCM_FORMAT_MU_LAW);
+  _EXPORT_INT(m,"PCM_FORMAT_A_LAW",SND_PCM_FORMAT_A_LAW);
+  _EXPORT_INT(m,"PCM_FORMAT_IMA_ADPCM",SND_PCM_FORMAT_IMA_ADPCM);
+  _EXPORT_INT(m,"PCM_FORMAT_MPEG",SND_PCM_FORMAT_MPEG);
+  _EXPORT_INT(m,"PCM_FORMAT_GSM",SND_PCM_FORMAT_GSM);
+
+}
diff --git a/playbacktest.py b/playbacktest.py
new file mode 100644
index 0000000..acc86f0
--- /dev/null
+++ b/playbacktest.py
@@ -0,0 +1,34 @@
+## recordtest.py
+##
+## This is an example of a simple sound playback script.
+##
+## The script opens an ALSA pcm for sound playback. Set
+## various attributes of the device. It then reads data
+## from stdin and writes it to the device.
+##
+## To test it out do the following:
+## python recordtest.py > out.raw # talk to the microphone
+## python playbacktest.py < out.raw
+
+import alsaaudio
+import sys
+import time
+
+# Open the device in playback mode. 
+out = alsaaudio.openpcm(alsaaudio.PCM_PLAYBACK)
+
+# Set attributes: Mono, 8000 Hz, 16 bit little endian frames
+out.setchannels(1)
+out.setrate(8000)
+out.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+
+# The period size controls the internal number of frames per period.
+# The significance of this parameter is documented in the ALSA api.
+out.setperiodsize(160)
+
+loops = 10000
+while loops > 0:
+  loops -= 1
+  # Read data from stdin
+  data = sys.stdin.read(320)
+  out.write(data)
diff --git a/recordtest.py b/recordtest.py
new file mode 100644
index 0000000..5a47e57
--- /dev/null
+++ b/recordtest.py
@@ -0,0 +1,45 @@
+## recordtest.py
+##
+## This is an example of a simple sound capture script.
+##
+## The script opens an ALSA pcm forsound capture. Set
+## various attributes of the capture, and reads in a loop,
+## writing the data to standard out.
+##
+## To test it out do the following:
+## python recordtest.py > out.raw # talk to the microphone
+## aplay -r 8000 -f S16_LE -c 1 out.raw
+
+import alsaaudio
+import sys
+import time
+
+# Open the device in nonblocking capture mode. The last argument could
+# just as well have been zero for blocking mode. Then we could have
+# left out the sleep call in the bottom of the loop
+inp = alsaaudio.openpcm(alsaaudio.PCM_CAPTURE,alsaaudio.PCM_NONBLOCK)
+
+# Set attributes: Mono, 8000 Hz, 16 bit little endian samples
+inp.setchannels(1)
+inp.setrate(8000)
+inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+
+# The period size controls the internal number of frames per period.
+# The significance of this parameter is documented in the ALSA api.
+# For our purposes, it is suficcient to know that reads from the device
+# will return this many frames. Each frame being 2 bytes long.
+# This means that the reads below will return either 320 bytes of data
+# or 0 bytes of data. The latter is possible because we are in nonblocking
+# mode.
+inp.setperiodsize(160)
+
+loops = 10000
+while loops > 0:
+  loops -= 1
+  # Read data from device
+  l,data = inp.read()
+  
+  if l:
+    # actual data read. Write it to stdout
+    sys.stdout.write(data)
+  time.sleep(.001)
diff --git a/setup.py b/setup.py
new file mode 100644
index 0000000..d2534a7
--- /dev/null
+++ b/setup.py
@@ -0,0 +1,14 @@
+from distutils.core import setup
+from distutils.extension import Extension
+
+setup(
+    name = "alsaaudio",
+    version = "0.1",
+    description = "alsa bindings",
+    author = "Casper Wilstrup",
+    author_email="cwi@unispeed.com",
+    ext_modules=[Extension("alsaaudio",["alsaaudio.c"],libraries=['asound'])
+                 ]
+    )
+    
+